This is the entry i use to
place #31# infront of a phonenumber
exten =
_00XX,2,Dial(SIP/SIPTrunk/#31#${EXTEN},55,o)
//Michael
-Ursprungligt meddelande-
Från:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] För Alyed Tzompa
Skickat: den 9 juli 2006 07:23
I just picked up a second hand packet8 dta310 model from a local
thrift shop. It has no power supply with it. It says it needs 12v
and 0.6a. I have a spare power supplies that might fit this
description.
What polarity does this unit require? positive tip? negative tip? If
you don't
Hi all,
I want to setup asterisk box to do the following jobs.
1- 100 cuncurent calls
2- 1000 User Registration
3- MySQL Realtim
4- PerlAGI
Here is my question could u please reply it:
1- No RTP only singnaling, Is it possible?
Ans:
2- How much RAM?
Ans:
3- How much bandhwidth per month with
You might want to look around the wiki (www.voip-info.org) as most of
your questions have answers there.
I want to setup asterisk box to do the following jobs.
1- 100 cuncurent calls
2- 1000 User Registration
3- MySQL Realtim
4- PerlAGI
Here is my question could u please reply it:
1- No RTP
On 7/9/06, Ryder Brook [EMAIL PROTECTED] wrote:
I have 2 POTs line coming into Asterisk. We have callerid feature from
Verizon on one of the lines.
What interface are the lines connected to?
I am not able to track any CallerID coming in, in the log. I am pretty green
with asterisk, and it's
I have 2 POTs line coming into Asterisk. We have callerid feature from
Verizon on one of the lines.
I am not able to track any CallerID coming in, in the log. I am pretty
green with asterisk, and it's not clear if I have to activate for
CallerID in the dialplan. The voicemail keeps saying
Hi
It looks like the X101P clones I bought from eBay are dogs, so I'll look
into buying some FXO-SIP box instead. Hopefully, I won't have the same
problems with static, or caller ID and call termination not being detected.
Still, considering the number of people having similar problems with
On Sun, 2006-07-09 at 10:22 +0200, Vincent Delporte wrote:
Still, considering the number of people having similar problems with those
cards, I was wondering what the problem is. Is it because the hardware, no
matter what is advertised, is actually not identical from card to card so
the
Michiel van Baak wrote:
If you buy a model without the spare in it's name, you
have the license to use them right ?
To use them with a CCM or CCME, yes :-)
How about secondhand phones you get from ebay ?
Is my cisco smartnet account enough to run the phone legally
? It's not a spare model
Tzafrir, are you trying to tell me that I can realy do double on the
intel becuase the second CPU will do it?
On 7/7/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Jul 06, 2006 at 03:32:04PM -0400, C F wrote:
I have recently build 2 machines, one with an Intel Pentium Dual Core
CPU, and
It looks like the X101P clones I bought from eBay are dogs, so I'll
look into buying some FXO-SIP box instead. Hopefully, I won't have the
same problems with static, or caller ID and call termination not being
detected.
Still, considering the number of people having similar problems with
Florian Overkamp wrote:
Michiel van Baak wrote:
If you buy a model without the spare in it's name, you
have the license to use them right ?
To use them with a CCM or CCME, yes :-)
How about secondhand phones you get from ebay ?
Is my cisco smartnet account enough to run the phone legally
?
Attilla De Groot wrote:
Hi all,
I have two pda's and I want to be able to make calls, but I need a
client for this. The only problem is Windows Mobile 5.0, I can't find
a freeware client for this, the only one is Sjphone. But this one is
still beta for windows mobile and it just doesn't
On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote:
Tzafrir, are you trying to tell me that I can realy do double on the
intel becuase the second CPU will do it?
In the ideal case you'll get double performance with two CPUs. In
theory.
A case of many concurrent calls is basically something
Hi,
Thank you for the suggestion.
I tried to use mISDN first, then CAPI and now I'm trying I4L.
As I'm using Debian, I can not load the FRITZ drivers. I got the source
from the official site and recompiled it, but there is a strange message in
the log and the capi drivers are not loaded.
Olivier Saulnier wrote:
Hello,
Do you know where i can download some rings for a PA1688 based Phone?
All rings on this link are not very nice...:
http://www.aredfox.com/edownloadsring.htm
Best regards,
Look at the technical documentation on the site, iirc the ringfiles are
encoded in a
Olivier wrote:
2006/7/6, Maxim Vexler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]:
NVFaxDetect does just that ;)
Do you think NVFaxDetect is reliable ?
Could you use it along a voicemail (I mean : someone having a single
extension for voice and fax call, forward all incoming calls to its
Hi,My situation is : I need to send fax from sip device attached fax over zap channel. Using G711, fax send ok, but is it posible to use t.38 protocol. Maybe someone can suggest me what software to use?
___
--Bandwidth and Colocation provided by
Picture this:
Exten = 100 #My Phone
Exten = 200 #MythPhone
Call comes in. Dialplan calls both extensions.
MythPhone is an add-on for MythTV,so when i receive a call,the CallerID
is flashed up on my TV.
I want to add another MythPhone to my other MythTV box upstairs.
Do i have to make a third
Marco Mouta wrote:
in your init-misdn.conf (or misdn.conf, not sure now...) you can
choose the MSNs for your incoming Ports or Outgoing ports,
msns=3223242,3223243,3223244
for example.
Then in your calls, just set the outgoing callerid for your trunk, to
one of them. Be aware that as far as
Picture this:
Exten = 100 #My Phone
Exten = 200 #MythPhone
Call comes in. Dialplan calls both extensions.
MythPhone is an add-on for MythTV,so when i receive a call,the CallerID
is flashed up on my TV.
I want to add another MythPhone to my other MythTV box upstairs.
Do i have to make a third
Fyi,
Double Intel Xeon 3Ghz performance below
g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex
ilbc
g723 - - - - - - - - - -
-
gsm - - 2 2 2 2 1 4 10 29
14
ulaw - 2 - 1 2 2 1 4 10 29
14
alaw - 2 1 - 2 2 1 4 10 29
14
g726 - 2 2 2 - 2 1 4 10 29
14
adpcm - 2 2 2 2 - 1 4 10 29
14
slin
On Sun, Jul 09, 2006 at 11:02:44AM -0400, Time Bandit wrote:
Picture this:
Exten = 100 #My Phone
Exten = 200 #MythPhone
Call comes in. Dialplan calls both extensions.
MythPhone is an add-on for MythTV,so when i receive a call,the CallerID
is flashed up on my TV.
I want to add another
Is there a Way I can have 2 phones, on the same extension (as in
Dial(SIP/phone1SIP/Phone2) ), whereby if one phone is in use, then the
other one will not be rung if called?
This is useful to me, in the situation where the phone system has a
queue and an agent will very often not be at his desk
Cisco 3660 with a PRI talking SIP to various Asterisk boxes
(not connected, separate PBXs) using ulaw all have issues with music on
hold being choppy. Normal voice and SIP (taking a call from the PRI,
placing a call or extension to extension calls) conversations are _perfect_ with no drop
snip
Another side issue... the Cisco phones have no way to remove the
installed firmware. Therefore, there is no way to legally sell a used
Cisco phone, period.
As a non Cisco user this whole discussion is enough to steer me away
from there VOIP products for good. I hate the idea that
At 12:00 09/07/2006 -0700, Rich Adamson [EMAIL PROTECTED] wrote:
Are some of the Ebay ad's misrepresented? Probably.
Thanks a lot for the info on the history of the FXO cards. Obviously, the
ones I bought aren't the good ones :-)
You're likely to become just about as frustrated with the
Good idea. Let's write some open-source firmware for Cisco phones.
What will you be contributing?
Martin Joseph wrote:
snip
Another side issue... the Cisco phones have no way to remove the
installed firmware. Therefore, there is no way to legally sell a used
Cisco phone, period.
As a
On Jul 9, 2006, at 12:11 PM, Jay Milk wrote:
Good idea. Let's write some open-source firmware for Cisco phones.
What will you be contributing?
Well, it looks like I already contributed the idea didn't I ;~)
I don't have any Cisco phones, so if you want to send be a couple I'll
take a
It looks like a good starting point would be here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup
On 7/9/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Is there a Way I can have 2 phones, on the same extension (as inDial(SIP/phone1SIP/Phone2) ), whereby if one phone is in use,
Hi everyone,
I know that functions like set_variable and get_variable (using php with
phpagi) only apply to the channel variable. What I need to do is reset a
global variable I have in our system. I have a script that is going to
determine when this will happen, but I just have to make it
Hi everyone,
I know that functions like set_variable and get_variable (using php with
phpagi) only apply to the channel variable. What I need to do is reset a
global variable I have in our system. I have a script that is going to
determine when this will happen, but I just have to make it
Olivier can you please do a cat /proc/cpuinfo and post it here? I
think you have a 64 bit cpu.
On 7/9/06, olivier.taylor [EMAIL PROTECTED] wrote:
Fyi,
Double Intel Xeon 3Ghz performance below
g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - -
Thanks for that Tzafrir. Why does it ignore the secend CPU?
BTW, on a side note on this topic, how can one calculate simultaneous
transcoded channels using show transalation?
In the case where it tells me 17 ms for encoding and 4 for decoding,
that gives me 21ms per channel, in what time frame
On Sun, 09 Jul 2006 19:17:36 +0100, Thomas Kenyon wrote:
Is there a Way I can have 2 phones, on the same extension (as in
Dial(SIP/phone1SIP/Phone2) ), whereby if one phone is in use, then the
other one will not be rung if called?
This is useful to me, in the situation where the phone system has
Skip local FXOs altogether. Setup an account with somone who provides DIDs via IP. Call forward your analog line to the IP based number. It will be absolutely painless compared to the troubles of small FXO interfaces.
Michael
On Sun, 09 Jul 2006 21:09:40 +0200, The Masked Cucumber wrote:
Part of a conversation with NuFone.
It is untrue, that they do not answer, but if than:
Quote:
3. change your attitude towards customers!!
No, if you don't like it, go use Vonage.
End of quote!
I had always problems with these people.
bye
Ronald
Ronald Wiplinger wrote:
Part of a conversation with NuFone.
It is untrue, that they do not answer, but if than:
Quote:
3. change your attitude towards customers!!
No, if you don't like it, go use Vonage.
End of quote!
I had always problems with these people.
bye
Ronald
i had a similar issue with the first branch of asterisk 1.2 and cheap
phones (tip-100 from tatung)
i'll suggest you to upgrade your asterisk box
are you using bristuff ?
try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1
lemme know
.mike
On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs
On Sun, 9 Jul 2006, Andrew D Kirch wrote:
To some extent I see your point and have been on the receiving end of
one of Jeremy's tirades.
I've since decided that NuFone is an interesting study in whether your
business can survive
with only clueful customers.
Some people are into SM I
On Thu, 2006-07-06 at 23:22 -0300, Fabio wrote:
are you using SIP reinvite ?
Proably not as I'm using t in Dial()s for call transfer.
post a bit more information (sip.conf)
[general]
context=sip
allowguest=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
domain=mydomain.com
domain=1.2.3.4
Hi all.
I am looking for a method to transfer a caller
on an existing session to another extension.
For example,
SIP/200 is currently talking with SIP/300.
I want to force user SIP/200 to be transfered to SIP/400.
(without myself being SIP/200. I am just an admin
at the asterisk console)
I upgraded one of the boxes to 1.2.9.1 and using native MOH I still get
it. I made sure to upgrade zaptel, etc as well.
I do have something of interest to note...
Placing the call on hold then taking it off hold and back on the music
is ok (doing that once it gets choppy) of course this is not
You will also want to add
no vad
to your dial-peer config to disable voice activity detection.
I do not think it will resolve your issue, but worth a shot.
-John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Bill Gibbs
Sent: Sunday, July 09,
Thanks to everyone.I had everything right except that rxgain and txgain were set to 0.I am actually embarassed to say that I spent most of Saturday getting lost and learning a lot and the stupid mistake was that the telephone that I was calling from has caller id blocked. Well, the only
Greetings,
Does anyone know where I can get app_txfax and app_rxfax
thatll work with spandsp 0.0.3pre22?
Ive tried the source in the snapshots/test-apps
directory. Ive also tried the files that came with t38bits.tgz. This is
what I get out of make:
make[1]: *** No rule to make
Kevin Smith wrote:
Hi everyone,
I know that functions like set_variable and get_variable (using php
with phpagi) only apply to the channel variable. What I need to do is
reset a global variable I have in our system. I have a script that is
going to determine when this will happen, but I just
The Masked Cucumber wrote:
At 12:00 09/07/2006 -0700, Rich Adamson [EMAIL PROTECTED] wrote:
Are some of the Ebay ad's misrepresented? Probably.
Thanks a lot for the info on the history of the FXO cards. Obviously,
the ones I bought aren't the good ones :-)
You're likely to become just
Dear Group,
I am having some problem with PRI, my calls randomly get disconnected
and after I am running Debug, I got the out from CLi screen...
Cli messages,
-- Executing Dial(Zap/31-1, zap/g1/100||rTt) in
new stack
-- Making new call for cr 32809
-- Requested transfer capability:
At 22:36 09/07/2006 -0700, Michael Graves [EMAIL PROTECTED] wrote:
Skip local FXOs altogether. Setup an account with somone who provides DIDs
via IP. Call forward your analog line to the IP based number. It will be
absolutely painless compared to the
troubles of small FXO interfaces.
I'll
50 matches
Mail list logo
